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audio merge (malc)
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1125 c046a42c-6fe2-441c-8c8c-71466251a162
This commit is contained in:
bellard
255
audio/mixeng.c
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255
audio/mixeng.c
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/*
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* QEMU Mixing engine
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*
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* Copyright (c) 2004 Vassili Karpov (malc)
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* Copyright (c) 1998 Fabrice Bellard
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include "vl.h"
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//#define DEBUG_FP
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#include "audio/mixeng.h"
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#define IN_T int8_t
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#define IN_MIN CHAR_MIN
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#define IN_MAX CHAR_MAX
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#define SIGNED
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#include "mixeng_template.h"
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#undef SIGNED
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#undef IN_MAX
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#undef IN_MIN
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#undef IN_T
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#define IN_T uint8_t
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#define IN_MIN 0
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#define IN_MAX UCHAR_MAX
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#include "mixeng_template.h"
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#undef IN_MAX
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#undef IN_MIN
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#undef IN_T
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#define IN_T int16_t
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#define IN_MIN SHRT_MIN
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#define IN_MAX SHRT_MAX
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#define SIGNED
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#include "mixeng_template.h"
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#undef SIGNED
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#undef IN_MAX
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#undef IN_MIN
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#undef IN_T
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#define IN_T uint16_t
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#define IN_MIN 0
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#define IN_MAX USHRT_MAX
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#include "mixeng_template.h"
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#undef IN_MAX
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#undef IN_MIN
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#undef IN_T
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t_sample *mixeng_conv[2][2][2] = {
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{
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{
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conv_uint8_t_to_mono,
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conv_uint16_t_to_mono
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},
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{
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conv_int8_t_to_mono,
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conv_int16_t_to_mono
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}
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},
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{
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{
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conv_uint8_t_to_stereo,
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conv_uint16_t_to_stereo
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},
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{
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conv_int8_t_to_stereo,
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conv_int16_t_to_stereo
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}
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}
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};
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f_sample *mixeng_clip[2][2][2] = {
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{
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{
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clip_uint8_t_from_mono,
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clip_uint16_t_from_mono
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},
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{
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clip_int8_t_from_mono,
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clip_int16_t_from_mono
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}
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},
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{
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{
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clip_uint8_t_from_stereo,
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clip_uint16_t_from_stereo
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},
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{
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clip_int8_t_from_stereo,
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clip_int16_t_from_stereo
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}
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}
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};
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/*
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* August 21, 1998
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* Copyright 1998 Fabrice Bellard.
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*
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* [Rewrote completly the code of Lance Norskog And Sundry
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* Contributors with a more efficient algorithm.]
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*
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* This source code is freely redistributable and may be used for
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* any purpose. This copyright notice must be maintained.
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* Lance Norskog And Sundry Contributors are not responsible for
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* the consequences of using this software.
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*/
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/*
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* Sound Tools rate change effect file.
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*/
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/*
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* Linear Interpolation.
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*
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* The use of fractional increment allows us to use no buffer. It
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* avoid the problems at the end of the buffer we had with the old
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* method which stored a possibly big buffer of size
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* lcm(in_rate,out_rate).
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*
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* Limited to 16 bit samples and sampling frequency <= 65535 Hz. If
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* the input & output frequencies are equal, a delay of one sample is
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* introduced. Limited to processing 32-bit count worth of samples.
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*
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* 1 << FRAC_BITS evaluating to zero in several places. Changed with
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* an (unsigned long) cast to make it safe. MarkMLl 2/1/99
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*/
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/* Private data */
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typedef struct ratestuff {
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uint64_t opos;
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uint64_t opos_inc;
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uint32_t ipos; /* position in the input stream (integer) */
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st_sample_t ilast; /* last sample in the input stream */
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} *rate_t;
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/*
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* Prepare processing.
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*/
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void *st_rate_start (int inrate, int outrate)
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{
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rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
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if (!rate) {
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exit (EXIT_FAILURE);
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}
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if (inrate == outrate) {
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// exit (EXIT_FAILURE);
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}
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if (inrate >= 65535 || outrate >= 65535) {
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// exit (EXIT_FAILURE);
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}
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rate->opos = 0;
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/* increment */
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rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate;
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rate->ipos = 0;
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rate->ilast.l = 0;
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rate->ilast.r = 0;
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return rate;
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}
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/*
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* Processed signed long samples from ibuf to obuf.
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* Return number of samples processed.
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*/
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void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
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int *isamp, int *osamp)
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{
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rate_t rate = (rate_t) opaque;
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st_sample_t *istart, *iend;
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st_sample_t *ostart, *oend;
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st_sample_t ilast, icur, out;
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int64_t t;
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ilast = rate->ilast;
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istart = ibuf;
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iend = ibuf + *isamp;
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ostart = obuf;
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oend = obuf + *osamp;
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if (rate->opos_inc == 1ULL << 32) {
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int i, n = *isamp > *osamp ? *osamp : *isamp;
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for (i = 0; i < n; i++) {
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obuf[i].l += ibuf[i].r;
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obuf[i].r += ibuf[i].r;
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}
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*isamp = n;
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*osamp = n;
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return;
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}
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while (obuf < oend) {
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/* Safety catch to make sure we have input samples. */
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if (ibuf >= iend)
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break;
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/* read as many input samples so that ipos > opos */
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while (rate->ipos <= (rate->opos >> 32)) {
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ilast = *ibuf++;
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rate->ipos++;
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/* See if we finished the input buffer yet */
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if (ibuf >= iend) goto the_end;
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}
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icur = *ibuf;
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/* interpolate */
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t = rate->opos & 0xffffffff;
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out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX;
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out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX;
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/* output sample & increment position */
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#if 0
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*obuf++ = out;
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#else
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obuf->l += out.l;
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obuf->r += out.r;
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obuf += 1;
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#endif
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rate->opos += rate->opos_inc;
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}
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the_end:
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*isamp = ibuf - istart;
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*osamp = obuf - ostart;
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rate->ilast = ilast;
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}
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void st_rate_stop (void *opaque)
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{
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qemu_free (opaque);
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}
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